Welltech LP-399 issues with Asterisk Now- Urgent Help Please

Hello everyone !

I`m new on the forum and sorry for my bad english.

One week ago i buyed two IP LAN Phones Welltech LP-399 to test the Asterisk Now solution because we need it to extend our company VoIP telephony on our network because in some locations we have wireless conection up to 15 km and cellphones are not working ( uncovered area because of geographical position ).
Thats way we networked in almost 60 km with wireless network ( transparent wireless networking / relaying / bridged .....).In the client points of the network we want to put some VoIP LAN phones and an Asterisk at the Central Building , thats why we buyed for test purposes this two phones WELLTECH LP-399.
The problems is regarding the codecs ( i supose ).

After configuration , Ive dialed the other phone , its ringing but when I`m answering i cant here anything ( me or the other person ).

The codecs used are :
G.711 u-law
G.711 a-law
G.726 - 16
G.726 - 24
G.726 - 32
G.726 - 40

RTP Packet Lenght is set ass fallowing
G.711 & G.729 = 10ms
iLBC = 20ms

Voice VAD = off
On the SIP Codecs i have let only the first codec to G.729 anthe the other 8 Codecs prority i set them to Not Used.

The Realm 1 is set as fallowing :

Display name = Aygun
User name = 6000
Register name = 6000
Register password = 1234
Domain server =
Proxy server =
Outbound proxy =

Subscribe to MWI = off

( note = where is the asterisk now ip )

SIP port is set to 5060
RTP port is set to 60000

The phone has two lan ports
One for WLAN and one for LAN configured as NAT ( i have two options , Bridge and NAT ) .
Its ok if i set it to NAT ? ( iam asking because i`m new to the voip stuff and i do not know very good some details ).

So let`s see the settingh of LAN port :
Ip =
subnet =
Dhcp server = off

Wan interface has fallowing settings:

LAN mode NAT

IP type = fixed
Hostname Aygun


Codec id settings are as fallowing

G.726 16 ID = 23
G.726 24 ID = 22
G.726 32 ID = 2
G.726 40 ID = 21
RFC 2833 ID = 101
iLBC ID = 97

DTMF Seetings set to RFC 2833

Rport = ON

Hold by RFC = OFF
Voice QoS (Diff-Serv) = 40
SIP QoS (Diff-Serv) = 40
SIP Expire Time = 60

The server is working fine because i have test it with Xlite VoIP softphone and I can talk an do what i whant to do.

Please can some one help me ?

Best regards to all of you.


Problem SOLVED !!!

In Asterisk NOW webpage at Configuration do you have ( Show Advanced Options ) .
After you click that button in the top right of the window in drop down meniu will apear, select Global SIP Seetings and at the bottom of the page do you have the last section named Codecs. In here you just edit the allowed codecs and add the codecs supported by your device.
Click save.

Activate Cnages and try again.

That`s it.

Have a nice day.